EP2847760B1

EP2847760B1 - Error-signal content controlled adaptation of secondary and leakage path models in noise-canceling personal audio devices

Info

Publication number: EP2847760B1
Application number: EP13721165.2A
Authority: EP
Inventors: Jeffrey Alderson; Jon D. Hendrix; Yang Lu
Original assignee: Cirrus Logic Inc
Current assignee: Cirrus Logic Inc
Priority date: 2012-05-10
Filing date: 2013-04-18
Publication date: 2021-06-02
Anticipated expiration: 2033-04-18
Also published as: CN104303228A; JP2015517683A; US9076427B2; WO2013169454A2; WO2013169454A4; KR102031536B1; WO2013169454A3; JP6305395B2; EP2847760A2; IN2014KN02311A; KR20150005714A; CN104303228B; US20130301849A1

Description

FIELD OF THE INVENTION

The present invention relates generally to personal audio devices such as wireless telephones that include adaptive noise cancellation (ANC), and more specifically, to control of ANC in a personal audio device that uses a measure of error signal content to control adaptation of secondary and leakage path estimates.

BACKGROUND OF THE INVENTION

Wireless telephones, such as mobile/cellular telephones, cordless telephones, and other consumer audio devices, such as MP3 players, are in widespread use. Performance of such devices with respect to intelligibility can be improved by providing noise-canceling using a microphone to measure ambient acoustic events and then using signal processing to insert an anti-noise signal into the output of the device to cancel the ambient acoustic events.
Noise-canceling operation can be improved by measuring the transducer output of a device to determine the effectiveness of the noise-canceling using an error microphone. The measured output of the transducer is ideally the source audio, e.g., downlink audio in a telephone and/or playback audio in either a dedicated audio player or a telephone, since the noise-canceling signal(s) are ideally canceled by the ambient noise at the location of the transducer. To remove the source audio from the error microphone signal, the secondary path from the transducer through the error microphone can be estimated and used to filter the source audio to the correct phase and amplitude for subtraction from the error microphone signal. Similarly, ANC performance can be improved by modeling the leakage path from the transducer to the reference microphone. However, when source audio is absent, the secondary path estimate and leakage path estimate cannot typically be updated. Further, when source audio is low in amplitude, the secondary path estimate and leakage path estimate may not be accurately updated, as the error microphone signal and/or the reference microphone signal may be dominated by other sounds.
Therefore, it would be desirable to provide a personal audio device, including wireless telephones, that provides noise cancellation using a secondary path estimate and/or leakage path estimates to remove the output of the transducer from error and reference signals, respectively, and that can determine whether or not to adapt the secondary path and leakage path estimates.
U.S. Patent Application Publication No. 2010/0061564 A1 discloses an adaptive, feed-forward, ambient noise-reduction system. The system includes a reference microphone for generating first electrical signals representing incoming ambient noise, and a connection path including a circuit for inverting these signals and applying them to a loudspeaker directed into the ear of a user. The system also includes an error microphone for generating second electrical signals representative of sound (including that generated by the loudspeaker in response to the inverted first electrical signals) approaching the user's ear. An adaptive electronic filter is provided in the connection path, together with a controller for automatically adjusting one or more characteristics of the filter in response to the first and second electrical signals. The system is configured to constrain the operation of the adaptive filter such that it always conforms to one of a predetermined family of filter responses, thereby restricting the filter to operation within a predetermined and limited set of amplitude and phase characteristics.
U.S. Patent Application Publication No. 2011/0142247 A1 teaches a method for adaptive control and equalization of electroacoustic channels. An electroacoustic channel soundfield is altered. An audio signal is applied by an electromechanical transducer to an acoustic space, causing air pressure changes therein. Another audio signal is obtained by a second electromechanical transducer, responsive to air pressure changes in the acoustic space. A transfer function estimate of the electroacoustic channel is established, responsive to the second audio signal and part of the first audio signal. The transfer function estimate is derived to be adaptive to temporal variations in the electroacoustic channel transfer function. Filters are obtained with transfer functions based on the transfer function estimate. Part of the first audio signal is filtered therewith.

DISCLOSURE OF THE INVENTION

The invention is defined in claims 1, 10, and 11, respectively. Particular embodiments are set out in the dependent claims.
In particular, the
above-stated objective of providing a personal audio device providing noise-cancelling including a secondary path and/or leakage path estimate that are adapted when sufficient source audio magnitude relative to ambient sounds is detected, is accomplished in a personal audio device, a method of operation, and an integrated circuit.
The personal audio device includes an output transducer for reproducing an audio signal that includes both source audio for providing to a listener and an anti-noise signal for countering the effects of ambient audio sounds in an acoustic output of the transducer. A microphone provides a measurement of ambient sounds, but that contains a component of source audio due to the transducer output. The personal audio device further includes an adaptive noise-canceling (ANC) processing circuit within the housing for adaptively generating an anti-noise signal from the at least one microphone signal such that the anti-noise signal causes substantial cancellation of the ambient audio sounds. The ANC processing circuit controls adaptation of an adaptive filter by compensating for the electro-acoustical path from the output of the processing circuit through the transducer into the at least one microphone, so that the component of the output of the at least one microphone can be corrected to remove components of source audio due to the transducer output. The ANC processing circuit permits the adaptive filter to adapt only when the content of the at least one microphone signal due to the source audio present in the transducer output relative to the microphone signal content due to the ambient audio is greater than a threshold, in order to properly model the acoustic and electrical paths.
The foregoing and other objectives, features, and advantages of the invention will be apparent from the following, more particular, description of the preferred embodiment of the invention, as illustrated in the accompanying drawings.

DESCRIPTION OF THE DRAWINGS

Figure 1A is an illustration of a wireless telephone 10 coupled to an earbud EB, which is an example of a personal audio device in which the techniques disclosed herein can be implemented.
Figure 1B is an illustration of electrical and acoustical signal paths in Figure 1A.
Figure 2 is a block diagram of circuits within wireless telephone 10.
Figure 3 is a block diagram depicting one example of an implementation of ANC circuit 30 of CODEC integrated circuit 20 of Figure 2.
Figure 4 is a block diagram depicting signal processing circuits and functional blocks within CODEC integrated circuit 20.

BEST MODE FOR CARRYING OUT THE INVENTION

The present invention encompasses noise-canceling techniques and circuits that can be implemented in a personal audio device, such as a wireless telephone. The personal audio device includes an adaptive noise canceling (ANC) circuit that measures the ambient acoustic environment and generates a signal that is injected into the speaker (or other transducer) output to cancel ambient acoustic events. A reference microphone is provided to measure the ambient acoustic environment, and an error microphone is included to measure the ambient audio and transducer output at the transducer, thus giving an indication of the effectiveness of the noise cancelation. A secondary path estimating adaptive filter is used to remove the playback audio from the error microphone signal, in order to generate an error signal. A leakage path estimating adaptive filter is used to remove the playback audio from the reference microphone signal to generate a leakage-corrected reference signal. However, depending on the relative amount of the transducer output relative to the ambient audio present in the error microphone signal, the secondary path estimate and leakage path estimate may not be updated properly. Therefore, update of the secondary path estimate and leakage path estimate is halted or otherwise managed when the relative amount of ambient audio to transducer output source audio content present in the error microphone signal exceeds a threshold.
Figure 1A shows a wireless telephone 10 proximate to a human ear 5. Illustrated wireless telephone 10 is an example of a device in which the techniques herein may be employed, but it is understood that not all of the elements or configurations illustrated in wireless telephone 10, or in the circuits depicted in subsequent illustrations, are required. Wireless telephone 10 is connected to an earbud EB by a wired or wireless connection, e.g., a BLUETOOTH™ connection (BLUETOOTH is a trademark of Bluetooth SIG, Inc.). Earbud EB has a transducer, such as speaker SPKR, which reproduces source audio including distant speech received from wireless telephone 10, ringtones, stored audio program material, and injection of near-end speech (i.e., the speech of the user of wireless telephone 10). The source audio also includes any other audio that wireless telephone 10 is required to reproduce, such as source audio from web-pages or other network communications received by wireless telephone 10 and audio indications such as battery low and other system event notifications. A reference microphone R is provided on a surface of a housing of earbud EB for measuring the ambient acoustic environment. Another microphone, error microphone E, is provided in order to further improve the ANC operation by providing a measure of the ambient audio combined with the audio reproduced by speaker SPKR close to ear 5, when earbud EB is inserted in the outer portion of ear 5. While the illustrated example shows an earbud implementation of a noise-canceling system, the techniques disclosed herein can also be implemented in a wireless telephone or other personal audio device, in which the output transducer and reference/error microphones are all provided on a housing of the wireless telephone or other personal audio device.
Wireless telephone 10 includes adaptive noise canceling (ANC) circuits and features that inject an anti-noise signal into speaker SPKR to improve intelligibility of the distant speech and other audio reproduced by speaker SPKR. Exemplary circuit 14 within wireless telephone 10 includes an audio CODEC integrated circuit 20 that receives the signals from reference microphone R, near-speech microphone NS, and error microphone E and interfaces with other integrated circuits such as an RF integrated circuit 12 containing the wireless telephone transceiver. In other embodiments of the invention, the circuits and techniques disclosed herein may be incorporated in a single integrated circuit that contains control circuits and other functionality for implementing the entirety of the personal audio device, such as an MP3 player-on-a-chip integrated circuit. Alternatively, the ANC circuits may be included within a housing of earbud EB or in a module located along a wired connection between wireless telephone 10 and earbud EB. For the purposes of illustration, the ANC circuits will be described as provided within wireless telephone 10, but the above variations are understandable by a person of ordinary skill in the art and the consequent signals that are required between earbud EB, wireless telephone 10 and a third module, if required, can be easily determined for those variations. The near-speech microphone NS is provided at a housing of wireless telephone 10 to capture near-end speech, which is transmitted from wireless telephone 10 to the other conversation participants). Alternatively, near-speech microphone NS may be provided on the outer surface of a housing of earbud EB, or on a boom (earpiece microphone extension) affixed to earbud EB.
Figure 1B shows a simplified schematic diagram of an audio CODEC integrated circuit 20 that includes ANC processing, as coupled to reference microphone R, which provides a measurement of ambient audio sounds Ambient that is filtered by the ANC processing circuits within audio CODEC integrated circuit 20. Audio CODEC integrated circuit 20 generates an output that is amplified by an amplifier A1 and is provided to speaker SPKR. Audio CODEC integrated circuit 20 receives the signals (wired or wireless depending on the particular configuration) from reference microphone R, near-speech microphone NS and error microphone E and interfaces with other integrated circuits such as an RF integrated circuit 12 containing the wireless telephone transceiver. In other configurations, the circuits and techniques disclosed herein may be incorporated in a single integrated circuit that contains control circuits and other functionality for implementing the entirety of the personal audio device, such as an MP3 player-on-a-chip integrated circuit. Alternatively, multiple integrated circuits may be used, for example, when a wireless connection is provided from earbud EB to wireless telephone 10 and/or when some or all of the ANC processing is performed within earbud EB or a module disposed along a cable connecting wireless telephone 10 to earbud EB.
In general, the ANC techniques illustrated herein measure ambient acoustic events (as opposed to the output of speaker SPKR and/or the near-end speech) impinging on reference microphone R, and also measure the same ambient acoustic events impinging on error microphone E. The ANC processing circuits of illustrated wireless telephone 10 adapt an anti-noise signal generated from the output of reference microphone R to have a characteristic that minimizes the amplitude of the ambient acoustic events at error microphone E. Since acoustic path P(z) extends from reference microphone R to error microphone E, the ANC circuits are essentially estimating acoustic path P(z) combined with removing effects of an electro-acoustic path S(z) that represents the response of the audio output circuits of CODEC IC 20 and the acoustic/electric transfer function of speaker SPKR. The estimated response includes the coupling between speaker SPKR and error microphone E in the particular acoustic environment which is affected by the proximity and structure of ear 5 and other physical objects and human head structures that may be in proximity to earbud EB. Leakage, i.e., acoustic coupling, between speaker SPKR and reference microphone R can cause error in the anti-noise signal generated by the ANC circuits within CODEC IC 20. In particular, desired downlink speech and other internal audio intended for reproduction by speaker SPKR can be partially canceled due to the leakage path L(z) between speaker SPKR and reference microphone R. Since audio measured by reference microphone R is considered to be ambient audio that generally should be canceled, leakage path L(z) represents the portion of the downlink speech and other internal audio that is present in the reference microphone signal and causes the above-described erroneous operation. Therefore, the ANC circuits within CODEC IC 20 include leakage-path modeling circuits that compensate for the presence of leakage path L(z). While the illustrated wireless telephone 10 includes a two microphone ANC system with a third near-speech microphone NS, a system may be constructed that does not include separate error and reference microphones. Alternatively, when near-speech microphone NS is located proximate to speaker SPKR and error microphone E, near-speech microphone NS may be used to perform the function of the reference microphone R. Also, in personal audio devices designed only for audio playback, near-speech microphone NS will generally not be included, and the near-speech signal paths in the circuits described in further detail below can be omitted.
Referring now to Figure 2, circuits within wireless telephone 10 are shown in a block diagram. CODEC integrated circuit 20 includes an analog-to-digital converter (ADC) 21A for receiving the reference microphone signal and generating a digital representation ref of the reference microphone signal, an ADC 21B for receiving the error microphone signal and generating a digital representation err of the error microphone signal, and an ADC 21C for receiving the near-speech microphone signal and generating a digital representation of near-speech microphone signal ns. CODEC IC 20 generates an output for driving speaker SPKR from an amplifier A1, which amplifies the output of a digital-to-analog converter (DAC) 23 that receives the output of a combiner 26. Combiner 26 combines audio signals ia from internal audio sources 24, the anti-noise signal anti-noise generated by ANC circuit 30, which by convention has the same polarity as the noise in reference microphone signal ref and is therefore subtracted by combiner 26, a portion of near-speech signal ns so that the user of wireless telephone 10 hears their own voice in proper relation to downlink speech ds, which is received from radio frequency (RF) integrated circuit 22. In accordance with an embodiment of the present invention, downlink speech ds is provided to ANC circuit 30. Combined downlink speech ds and internal audio ia forming source audio (ds+ia) is provided to combiner 26, so that source audio (ds+ia) is always present to estimate acoustic path S(z) with a secondary path adaptive filter within ANC circuit 30. Near-speech signal ns is also provided to RF integrated circuit 22 and is transmitted as uplink speech to the service provider via antenna ANT.
Figure 3 shows one example of details of ANC circuit 30 that can be used to implement ANC circuit 30 of Figure 2. A combiner 36A removes an estimated leakage signal from reference microphone signal ref, which in the example is provided by a leakage-path adaptive filter 34C having a response LE(z) that models leakage path L(z). Combiner 36A generates a leakage-corrected reference microphone signal ref. An adaptive filter 32 receives leakage-corrected reference microphone signal ref and under ideal circumstances, adapts its transfer function W(z) to be P(z)/S(z) to generate the anti-noise signal anti-noise, which is provided to an output combiner that combines the anti-noise signal with the audio to be reproduced by speaker SPKR, as exemplified by combiner 26 of Figure 2. The coefficients of adaptive filter 32 are controlled by a W coefficient control block 31 that uses a correlation of two signals to determine the response of adaptive filter 32, which generally minimizes the error, in a least-mean squares sense, between those components of leakage-corrected reference microphone signal ref present in error microphone signal err. The signals processed by W coefficient control block 31 are the leakage-corrected reference microphone signal ref shaped by a copy of an estimate of the response of path S(z) (i.e., response SE_COPY(z)) provided by filter 34B and another signal that includes error microphone signal err. By transforming leakage-corrected reference microphone signal ref with a copy of the estimate of the response of path S(z), response SE_COPY(z), and minimizing error microphone signal err after removing components of error microphone signal err due to playback of source audio, adaptive filter 32 adapts to the desired response of P(z)/S(z).
In addition to error microphone signal err, the other signal processed along with the output of filter 34B by W coefficient control block 31 includes an inverted amount of the source audio (ds + ia) including downlink audio signal ds and internal audio ia. Source audio (ds+ia) is processed by a filter 34A having response SE(z), of which response SE_COPY(z) is a copy. Filter 34B is not an adaptive filter, per se, but has an adjustable response that is tuned to match the response of adaptive filter 34A, so that the response of filter 34B tracks the adapting of adaptive filter 34A. By injecting an inverted amount of source audio (ds + ia) that has been filtered by response SE(z), adaptive filter 32 is prevented from adapting to the relatively large amount of source audio (ds + ia) present in error microphone signal err. By transforming the inverted copy of downlink audio signal ds and internal audio ia with the estimate of the response of path S(z), the source audio (ds + ia) that is removed from error microphone signal err before processing should match the expected version of downlink audio signal ds and internal audio ia reproduced at error microphone signal err. The source audio (ds + ia) matches the amount of source audio (ds + ia) present in error microphone signal err because the electrical and acoustical path of S(z) is the path taken by source audio (ds + ia) to arrive at error microphone E.
To implement the above, adaptive filter 34A has coefficients controlled by SE coefficient control block 33A, which processes the source audio (ds+ia) and error microphone signal err after removal, by a combiner 36B, of the above-described filtered downlink audio signal ds and internal audio ia, that has been filtered by adaptive filter 34A to represent the expected source audio delivered to error microphone E. Adaptive filter 34A is thereby adapted to generate an error signal e from downlink audio signal ds and internal audio ia, that when subtracted from error microphone signal err, contains the content of error microphone signal err that is not due to source audio (ds+ia). Similarly, LE coefficient control 33B also is adapted to minimize the components of source audio (ds+ia) present in leakage-corrected reference microphone signal ref, by adapting to generate an output that represents the source audio (ds+ia) present in reference microphone signal ref. However, if downlink audio signal ds and internal audio ia are both absent or low in amplitude, the content of error microphone signal err and reference microphone signal ref will primarily consist of ambient sounds, which may not be suitable for adapting response SE(z) and response LE(z). Therefore, error microphone signal err may have sufficient amplitude, and yet be unsuitable in content to be useful as a training signal for response SE(z). Similarly, reference microphone signal ref may not contain the proper content to train response LE(z). In ANC circuit 30, a source audio detector 35A detects whether sufficient source audio (ds + ia) is present, and a comparison block 39 updates the secondary path estimate and leakage path estimate if sufficient source audio (ds + ia) is present as indicated by the magnitude of control signal Source Level. The threshold applied to determine whether sufficient source audio (ds + ia) is present can be determined from a magnitude of reference microphone signal ref, as determined by a reference level detector 35B, and as indicated by the magnitude of control signal Reference Level. Comparison block 39 determines whether the magnitude of control signal Source Level is sufficiently great compared to the magnitude of control signal Reference Level and de-asserts control signal haltSE to permit SE coefficient control 33A to update response SE(z) only if sufficient source audio (ds + ia) is present. Similarly, comparison block 39 de-asserts control signal haltLE to permit LE coefficient control 33B to update response LE(z) only if sufficient source audio (ds + ia) is present and may apply the same criteria as for control signal haltSE, or a different threshold may be used. Level detector 35B includes both amplitude detection, and optionally filtering, to obtain the magnitude of reference microphone signal ref. In one exemplary implementation, reference level detector 35B uses a wideband root-mean-square (RMS) detector to determine the magnitude of the ambient sounds. In another example, reference level detector 35B includes a filter that filters reference microphone signal ref to select one or more frequency bands before making an RMS amplitude measurement, so that particular frequencies that will cause improper adaptation of response SE(z) and response LE(z) can be prevented from causing such a disruption, while other sources of ambient noise might be permitted while adapting response SE(z) and response LE(z).
An alternative to using source audio detector 35A to determine the relative amount of source audio (ds + ia) present in error microphone signal err, is to use a volume control signal Vol ctrl as an indication of the magnitude of source audio (ds + ia) being reproduced by speaker SPKR. Volume control signal Vol ctrl is applied to source audio (ds + ia) by a gain stage g1, which also controls the amount of source audio (ds + ia) provided to adaptive filter 34A and adaptive filter 34C. Additionally, whether volume control signal Vol ctrl or control signal Source Level is compared to the threshold provided by control signal Reference Level, the degree of coupling between the listener's ear and personal audio device 10 can be estimated by an ear pressure estimation block 38 to further refine the determination of whether response SE(z) and response LE(z) can be adapted. Ear pressure estimation block 38 generates an indication, control signal pressure, of the degree of coupling between the listener's ear and personal audio device 10. Comparison block 39 can then use control signal Pressure to reduce the threshold provided by control signal Reference Level, since a higher value of control signal Pressure generally indicates that the source audio present in the acoustic output of speaker SPKR is more effectively coupled to the listener's ear, and thus for a given level of source audio (ds + ia), the amount of source audio (ds + ia) heard by the listener is increased with respect to the level of ambient noise. Techniques for determining the degree of coupling between the listener's ear and personal audio device 10 that may be used to implement comparison block 39 are disclosed in U.S. Patent Application Publication US20120207317A1 entitled "EAR-COUPLING DETECTION AND ADJUSTMENT OF ADAPTIVE RESPONSE IN NOISE-CANCELING IN PERSONAL AUDIO DEVICES".
Referring now to Figure 4 , a block diagram of an ANC system is shown for implementing ANC techniques as depicted in Figure 3, and having a processing circuit 40 as may be implemented within CODEC integrated circuit 20 of Figure 2. Processing circuit 40 includes a processor core 42 coupled to a memory 44 in which program instructions are stored, the program instructions comprising a computer-program product that may implement some or all of the above-described ANC techniques, as well as implementing other signal processing algorithms. Optionally, a dedicated digital signal processing (DSP) logic 46 may be provided to implement a portion of, or alternatively all of, the ANC signal processing provided by processing circuit 40. Processing circuit 40 also includes ADCs 21A-21C, for receiving inputs from reference microphone R, error microphone E and near-speech microphone NS, respectively. DAC 23 and amplifier A1 are also provided by processing circuit 40 for providing the transducer output signal, including anti-noise as described above.
While the invention has been particularly shown and described with reference to the preferred embodiments thereof, it will be understood by those skilled in the art that the foregoing, as well as other changes in form and details may be made therein without departing from the scope of the invention, as defined by the following claims.

Claims

An integrated circuit for implementing at least a portion of a personal audio device (10), comprising:
an output adapted to provide an output signal to an output transducer (SPKR) including both source audio for playback to a listener and an anti-noise signal for countering the effects of ambient audio sounds in an acoustic output of the transducer (SPKR);

at least one microphone input adapted to receive at least one microphone signal indicative of the ambient audio sounds and that contains a component due to the acoustic output of the transducer (SPKR); and

a processing circuit (20, 30) configured to adaptively generate the anti-noise signal to reduce the presence of the ambient audio sounds heard by the listener, wherein the processing circuit (20, 30) implements an adaptive filter having a response that shapes the source audio and a combiner (36A) that removes the shaped source audio from the at least one microphone signal to provide a corrected microphone signal, wherein the processing circuit (20, 30) is configured to determine a relative magnitude of a source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal, wherein the processing circuit (20, 30) is configured to determine a degree of coupling between the transducer (SPKR) and an ear (5) of the listener, and wherein the processing circuit (20, 30) is configured to take action to prevent improper adaptation of the adaptive filter in response to determining that the relative magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal to the ambient audio sounds present in the at least one microphone signal indicates that the adaptive filter may not adapt properly;

characterised in that

the processing circuit (20, 30) is configured to adjust the determined relative magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal in conformity with the determined degree of coupling.
The integrated circuit of Claim 1, wherein the at least one microphone signal includes an error microphone signal (err) indicative of the ambient audio sounds and the acoustic output of the transducer (SPKR), wherein the adaptive filter is a secondary path adaptive filter (34A) that adapts to model a response of a secondary path taken by the source audio through the transducer (SPKR) and into the error microphone signal (err), and wherein an output of the secondary path adaptive filter (34A) is combined with the error microphone signal (err) to generate an error signal (e) indicative of the source audio component of the acoustic output of the transducer (SPKR).
The integrated circuit of Claim 2, wherein the at least one microphone signal includes a reference microphone signal (ref) indicative of the ambient audio sounds, and further comprising a leakage path adaptive filter (34C) that adapts to model a response of a leakage path taken by the source audio through the transducer (SPKR) and into the reference microphone signal (ref), and wherein an output of the leakage path adaptive filter (34C) is combined with the reference microphone signal (ref) to generate a leakage-corrected reference microphone signal (ref) from which the anti-noise signal is generated.
The integrated circuit of Claim 2, wherein the processing circuit (20, 30) is configured to compute a ratio of a first magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the error signal (e) relative to a second magnitude of the ambient audio sounds present in the error signal (e) and to compare the ratio to a threshold, wherein the processing circuit (20, 30) is further configured to halt adaptation of the secondary path adaptive filter (34A) in response to determining that the ratio is less than the threshold.
The integrated circuit of Claim 1, wherein the at least one microphone signal includes a reference microphone signal (ref) indicative of the ambient audio sounds, wherein the adaptive filter is a leakage path adaptive filter (34C) that adapts to model a response of a leakage path taken by the source audio through the transducer (SPKR) and into the reference microphone signal (ref), and wherein an output of the leakage path adaptive filter (34C) is combined with the reference microphone signal (ref) to generate a leakage-corrected reference microphone signal (ref) from which the anti-noise signal is generated.
The integrated circuit of Claim 1, wherein the processing circuit (20, 30) is configured to detect a magnitude of the source audio and to use the magnitude of the source audio to determine the magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal; or wherein the processing circuit (20, 30) is configured to use a volume control setting (Vol ctrl) applied as gain to the source audio to determine the magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal.
The integrated circuit of Claim 1, wherein the processing circuit (20, 30) is configured to detect a magnitude of the ambient sounds using the at least one microphone signal, and wherein the processing circuit (20, 30) is further configured to use the magnitude of the ambient audio sounds to determine the magnitude of the ambient audio sounds present in the at least one microphone signal.
The integrated circuit of Claim 7, wherein the processing circuit (20, 30) is configured to detect the magnitude of the ambient sounds by determining a wideband root-mean-square amplitude of the at least one microphone signal or by determining a root-mean-square amplitude of the at least one microphone signal in one or more predetermined frequency bands.
The integrated circuit of Claim 7, wherein the processing circuit (20, 30) is configured to detect a magnitude of the source audio and to compare the magnitude of the source audio to a magnitude of the at least one microphone signal to determine the relative magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal, and wherein the processing circuit (20, 30) is preferably further configured to adjust the comparing of the magnitude of the source audio to the magnitude of the at least one microphone signal by adjusting the magnitude of the at least one microphone signal that is compared to the magnitude of the at least one microphone signal in conformity with the determined degree of coupling.
A personal audio device, comprising:
a personal audio device housing;

an integrated circuit (20, 30) according to any one of Claims 1-9;

a transducer (SPKR) mounted on the housing and coupled to the output of the integrated circuit (20, 30); and

at least one microphone mounted on the housing and coupled to the at least one microphone input of the integrated circuit (20, 30).
A method of countering effects of ambient audio sounds by a personal audio device (10), the method comprising:
adaptively generating an anti-noise signal to reduce the presence of the ambient audio sounds heard by a listener;

combining the anti-noise signal with source audio;

providing a result of the combining to a transducer (SPKR);

measuring the ambient audio sounds and an acoustic output of the transducer (SPKR) with at least one microphone;

implementing an adaptive filter having a response that shapes the source audio and a combiner that removes the shaped source audio from at least one microphone signal to provide a corrected microphone signal to the at least one microphone;

determining a relative magnitude of a source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal;

determining a degree of coupling between the transducer (SPKR) and an ear (5) of the listener; and

taking action to prevent improper adaptation of the adaptive filter in response to determining that the relative magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal to the ambient audio sounds present in the at least one microphone signal indicates that the adaptive filter may not adapt properly;

characterised by

adjusting the determined relative magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal in conformity with the determined degree of coupling.
The method of Claim 11, wherein the at least one microphone signal includes an error microphone signal (err) provided by an error microphone (E) mounted on a housing proximate to the transducer (SPKR), wherein the adaptive filter is a secondary path adaptive filter (34A) that adapts to model a response of a secondary path taken by the source audio through the transducer (SPKR) and into the error microphone signal (err), and wherein the method further comprises combining an output of the secondary path adaptive filter (34A) with the error microphone signal (err) to generate an error signal (e) indicative of the source audio component of the acoustic output of the transducer (SPKR).
The method of Claim 12, wherein the determining comprises computing a ratio of a first magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the error signal (e) relative to a second magnitude of the ambient audio sounds present in the error signal (e) and comparing the ratio to a threshold, and wherein the taking action comprises halting adaptation of the secondary path adaptive filter (34A) in response to determining that the ratio is less than the threshold.
The method of Claim 11 or 12, wherein the at least one microphone signal further includes a reference microphone signal (ref) provided by a reference microphone (R) mounted on the housing for measuring the ambient audio sounds, and wherein the method further comprising:
generating a leakage correction signal using a leakage path adaptive filter (34C) that adapts to model a response of a leakage path taken by the source audio through the transducer (SPKR) and into the reference microphone signal (ref); and

combining the leakage correction signal with the reference microphone signal (ref) to generate a leakage-corrected reference microphone signal (ref) from which the anti-noise signal is generated.
The method of Claim 11, further comprising detecting a magnitude of the source audio, wherein the determining uses the detected magnitude of the source audio to determine the magnitude of the source audio component of acoustic output of the transducer (SPKR) present in the at least one microphone signal.
The method of Claim 11, wherein the determining uses a volume control setting applied as gain to the source audio to determine the magnitude of the source audio component of the acoustic output of the transducer (SPKR) present in the at least one microphone signal.
The method of Claim 11, further comprising detecting a magnitude of the ambient sounds using the at least one microphone signal, and wherein the determining uses the magnitude of the ambient audio sounds to determine the magnitude of the ambient audio sounds present in the at least one microphone signal.
The method of Claim 17, wherein the detecting detects the magnitude of the ambient sounds by determining a wideband root-mean-square amplitude of at least one microphone signal generated by the at least one microphone or by determining a root-mean-square amplitude of at least one microphone signal generated by the at least one microphone in one or more predetermined frequency bands.
The method of Claim 17, wherein the detecting detects a magnitude of the source audio and compares the magnitude of the source audio to a magnitude of the at least one microphone signal generated by the at least one microphone to determine the relative magnitude of the source audio component of the acoustic output of the transducer present in the at least one microphone signal and the ambient audio sounds present in the at least one microphone signal, and preferably further comprising adjusting the comparing of the magnitude of the source audio to a magnitude of the at least one microphone signal by adjusting the magnitude of the at least one microphone signal that is compared to the magnitude of the at least one microphone signal in conformity with the determined degree of coupling.